The present invention relates to a method and a device identifying an unknown system such as a transmission path, an acoustic coupling path or the like.
Echo cancellers, noise cancellers, howling cancellers, equalizers and the like are known as applications for unknown system identification with an adaptive filter. Here, using an acoustic echo canceller removing acoustic echo leaking into a microphone from a speaker in an acoustic coupling path as an example, a prior art is explained.
An echo canceller uses an adaptive filter having a larger number of tap coefficients than the impulse response length of an echo path, generates an echo replica corresponding to a transmission signal and reduces acoustic echo leaking into a microphone from a speaker in an acoustic coupling path. At this time, each tap coefficient of the adaptive filter is corrected based on correlation between a far-end signal and an error signal that is obtained by subtracting the echo replica from a mixed signal in which echoes and a near-end signal exist in mixture.
Following papers are known as typical coefficient adaptation algorithms for such an adaptive filter; "LMS Algorithm" (Proceedings of IEEE, pp.1692-1716, Vol.63-No.12, 1975; hereinafter referred to as Paper 1) and "Learning Identification Method; LIM" (IEEE Transactions on Automatic Control, pp.282-287, Vol.12-No.3, 1967; hereinafter referred to as Paper 2).
The impulse response length of the acoustic space where an acoustic echo canceller is actually used is dependent on physical dimensions of the acoustic space and a reflection factor of the wall. For example, assuming a teleconference room, the impulse response length of the room reaches 1,000 taps and sometimes several thousands taps. From the viewpoint of computation and hardware size, realization of such an echo canceller is extremely difficult in this case. Therefore, subband adaptive filters are proposed to solve problems such as increased amount of computation and the like. "IEEE SP Magazine" (pp.14-37, January in 1992; hereinafter referred to as Paper 3).
First, according to a method of Paper 3 shown in FIG. 11, an input signal is divided into a plurality of subbands with an analysis filter bank 3 and subband input signals are generated. This subband input signal is decimated by a factor of L.sub.1 in decimation circuits 50.sub.i (i=1, 2, . . . , K) and is supplied to the adaptive filters 61.sub.i (i=1, 2, . . . , K) that are independent each other. L.sub.i is usually set as L.sub.i =K.
On the other hand, an output of unknown system 2 to be identified, that is an echo of the echo canceller application, is also divided into a plurality of subbands with an analysis filter bank 4 having quite same characteristics as the analysis filter bank 3, becomes a subband echo and decimated by a factor of L.sub.i in the sampling circuits 51.sub.i (i=1, 2, . . . , k).
A subband echo replica is generated from the decimated subband input signal in an adaptive filter 61.sub.i. A subband error signal that is a difference between this subband echo replica and the decimated subband echo is generated. The adaptive filter 61.sub.i uses this subband error signal and performs coefficient update.
This subband error signal is interpolated by a factor of L.sub.i in the interpolation circuits 70.sub.i (i=1, 2, . . . , K), supplied to the synthesis filter bank 8, synthesized and transmitted to the output terminal 9. Accordingly if a subband error signal is small enough, that is, if a subband echo is suppressed enough in each subband, a signal obtained in the output terminal 9 becomes a full-band signal with a minimum residual echo.
There are various kinds of structures for the adaptive filter 61.sub.i, however, the most common structure is the FIR adaptive filter. (IEEE Transactions on Acoustics, Speech and Signal Processing, pp.768-781, Vol.27-No.6, 1979. hereinafter referred to as Paper 4).
FIG. 12 a block diagram of an FIR adaptive filter. A subband input signal from the decimation circuit 50.sub.i is supplied to the input terminal 610 and a subband reference signal from the decimation circuit 51.sub.i is supplied to the input terminal 620. Also, a signal obtained in the output terminal 630 is transmitted to the interpolation circuit 70.sub.i.
A signal supplied to input terminal 610 is supplied to a tapped delay line comprising a plurality of delay elements 611.sub.1, . . . , 611.sub.N-1 that generates a delay of one sampling period. A sample of an input signal supplied to the delay element 611.sub.1 is transferred to a delay element next to at every single clock. The output signal of each delay element 611.sub.i (i=1, 2, . . . , N-1) is supplied to the corresponding multiplier 613.sub.i+1, multiplied by a signal supplied from the corresponding coefficient generation circuit 612.sub.i+1. A signal is supplied to the multiplier 613.sub.i directly from the input terminal 610.
After summed up in adder 614 and subtracted from the subband reference signal in subtracter 616, the all output signals of multiplier 613.sub.i, . . . 613.sub.N are transmitted to the output terminal 630.
Assuming the LMS algorithm shown in Paper 1 as the coefficient adaption algorithm, a block diagram that shows configuration of the coefficient generation circuit 612.sub.i (i=1, 2, . . . , N) can be expressed as FIG. 13. A signal from the subtracter 616 in FIG. 12 is supplied to the input terminal 640 and a signal from the input terminal 610 in FIG. 12 is supplied to the input terminal 650 and a signal obtained in the output terminal 660 is transmitted to the multiplier 613.sub.i in FIG. 12. Also, a signal from the input terminal 640 and a signal from the input terminal 650 are multiplied in multiplier 641 and the result is supplied to adder 643. Multiplier 644 multiplies the signal supplied from the multiplier 641 by a constant .mu., and outputs the result to adder 643. On the other hand, adder 643 is supplied with an output of adder 643 itself delayed one sampling period by delay element 642. Accumulation of values supplied by multiplier 644 is carried out by a loop circuit consisting of delay element 642 and adder 643. Going through this loop circuit once is equivalent to carrying out a single coefficient update.
The number of taps N of the FIR adaptive filter that is explained with FIG. 12 must be equal to or greater than the corresponding impulse response length.
Generally, a subband acoustic echo lower band lasts longer than that in a high band. This is because the impulse response length of an acoustic echo is basically determined by a reflection and the reflection factor for a high band component is smaller than that for a low band resulting in shorter reverberation.
FIG. 14A, FIG. 14B, FIG. 14C and FIG. 14D show examples of a typical impulse response in each band for a 4-band acoustic echo. Assuming the impulse response length in band 1, . . . , band 4 are M1, M2, M3 and M4 respectively, M1.gtoreq.M2.gtoreq.M3.gtoreq.M4 is satisfied.
In the conventional examples explained above, the number of taps of adaptive filter 61.sub.i (i=1, 2, . . . , K) is always equal in all subbands and set equal to the number of taps necessary for the subband that requires the largest number.
By this reason, in a subband that needs a smaller number of taps, namely a high subband in general, the number of taps becomes too large, resulting in increased computation and larger convergence time due to interference by excess taps.